Unify (Siemens) OpenStage 15 and freePBX

Hi guys!

I have Unify OpenStage 15 on my lab and i would like to connect it to freePBX
Now i changed firmware to SIP.
I created extension using SIP driver and configured everythink in: SystemIdentity, SIP interface and Registrations sections.
On my phone appear error like: “No telephony possible (RF2)”
And my asterisk CLI show error: "[2023-04-20 23:30:33] WARNING[5425]: res_pjsip_registrar.c:1189 registrar_on_rx_request: Endpoint ‘anonymous’ ( has no configured AORs
What i’am doing bad?

I have successfully registered OpenScape Desk Phone CP200 in the past with FreePBX. I don’t know if the web gui of the OpenStage 15 is a different one but I had to set the “Terminal number” to the extension number. You can find the setting under System > System identity.

Thanks for reply. I had set Terminal number as extension to but phone doesnt register.
I suppose that my extension on the freepbx side is wrongly created maybe it needs some specific parameters for sip (as in case of cisco phones) isn’t it? or it is on default SIP driver ?

How are you expecting the Siemens to identify itself. Asterisk is basically saying that it doesn’t know who sent the registration request, and has matched it up against the anonymous one. You should probably turn off anonymous and guest options, as they shouldn’[t be needed for chan_pjsip and can cause security problems. This will not fix the problem here, but you will get a security violation, rather than failed to find AORs.

To see if there is any way of handling this from the Asterisk side, please provide a pjsip set logger on type capture of the REGISTER request, and point out the information in that which could be used to identify the sender.

We must start with the fact that I am not sure which driver to use pjsip or sip, nowhere in the case of OpenStage is written about it.

From Siemen’s point of view, they are both SIP drivers. chan_sip is no longer present in the development branch of Asterisk, so will not be in the upcoming release of Asterisk. That means you only really have one choice: chan_pjsip.

I can confirm it is definitely working with pjsip. Besides the logs as david mentioned can you also post the System identity, SIP interface and Registration settings of the phones web gui. If I remember correctly I had the same issue with the CP200 but it only was a small setting on the phone. I also have access to one CP200 right now and can compare.

Did you have any value in Terminal Name field? I think that was my problem in the past. This field needs to be empty. But I am not 100 percent sure.

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